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Discussion Starter #21
Here we go, more test results Wayne:

No 1 Sub behind TV.
No 2 Sub beside TV.
No 3 Main spk Large.
No 4 Main 120hz.
No 5 Main 80hz.
No 5 Main 40hz.

The sub I'm all ears as to which results you think looks better, 1 or 2, any advice where the BFD should improve?


Now I moved my main spk's around and found the best result was where they started from.

Cheers. Will
 

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I’d say the green trace looks best. It looks to me like you’d be getting really “bottom-heavy” sound with the sub, like maybe a car door closing sounding like a low-level distant explosion. If that’s the case, the BFD could tame the lowest frequencies. Also if you go with the green graph, some boost between 50-70 Hz would probably get audible results, especially with music.

I’d go for 120 Hz for the main speakers, but ONLY if you can set the sub for a lower frequency, like 80-90 Hz.

Regards,
Wayne
 

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Discussion Starter #23
I’d go for 120 Hz for the main speakers, but ONLY if you can set the sub for a lower frequency, like 80-90 Hz.

My PC sound card has a variable crossover from 50hz to 250hz, so I could set this to 120hz, then use the SVS crossover set to 80-90hz.

I'll give the BFD a whirl and see how I get on, I've nothing to loose......:T
 

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Discussion Starter #24
Hi again.
To add to confusion I have been carrying out test's using two different lap tops and a PC, the PC has a ASUS xonar 1.3 deluxe sound card.
The results on page one of this thread where taken using a Siemens L/T, the other results on page two using a Dell L/T. Now the results do vary slightly on these I guess this is down to the sound cards performance?
However the results on my PC are way different, can any one advise me as to where I could be going wrong please, I have checked all the settings on all L/P and PC and they seem to be all the same...:scratch:

PC results below
 

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Discussion Starter #25
I have just found that if I do not use a sound card calibrated file then the result is as below.
This again is a bit of a mystery as when calibrating the SC it is only out by 1dB at the most
 

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This again is a bit of a mystery as when calibrating the SC it is only out by 1dB at the most
Your soundcard may have shown only 1dB over most of the range, but if you look in the two graphs in your penultimate message, you can see in dotted black line that is the soundcard curve that it falls off at the bottom, and is in fact -80dB down at your left margin below 20Hz. So removing it from the configuration will have a huge effect on the displayed measure at the low end.

Now as to why your soundcard calibration starts falling off below 80Hz, I'm not sure. You mention that you have a variable crossover on the soundcard; perhaps you had it enabled when you generated the soundcard calibration file.

Bill
 

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Discussion Starter #27
Thanks Bill, I had my bass management. :rolleyesno:

I've managed to calibrate it now. But unfortunately my results have not improved that much, I guess my room sucks.

Below are some results from tonight, I have not run the Anti-mode as I wanted to see what the REW filters could achieve. The EQ result below only had two filters applied by REW.

I am going to have a go at using my DSP1124P this weekend to see if this will help, or at least to try ang get my head round it.

Whats the best plan off attack for using this:
Do I use the REW EQ suggested filters when I carry out the 'match response target', is there a limit to how many I should use, I guess fewer is better, but what if I used all twelve.
I've also read in the threads about not adding to much Gain above 3dB.....?

Any advice once again would be truly appreciated.

Man it's been one confusing last two weeks, It hasn't helped with my lap top's dying one by one!

Cheers.
Will
 

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Will, your target level is much too low, possibly because you did not go through the SPL meter calibration step so your measurements are being plotted at SPL values much higher than they actually are (at least I hope so, 125dB+ would be quite a loud measurement :))

After you have gone through the SPL meter calibration and made a measurement, open the Target Settings panel in the EQ window (click on the title bar) and make sure the crossover frequency ties up with what your sub is set to use. Then click on "Set Target Level" to get REW to adjust the level to tie up with the response. After doing that you can try "Match Response to Target" to see what filters REW suggests.

Your anti-mode plot is rather unusual though, it is normally very good at dealing with peaks so I suspect something did not go right in setting up the Anti-Mode.
 

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Discussion Starter #30
Your right John, it was out.

It's hard to get an accurate figure when the SPL meter jumps around so much.
Is it best to take the highest level reading as the max, or average the middle, as it can jump around from 72dB to 75dB, even more so when the anti mode is off?
Green Anti mode on...
 

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Is it best to take the highest level reading as the max, or average the middle, as it can jump around from 72dB to 75dB, even more so when the anti mode is off?
Take the average, the exact figure is not critical. Now that your levels are more sensible you can set the graph SPL axis to our preferred 45dB .. 105dB, having a consistent axis range makes it easier to visually interpret the graphs.

P.S. The anti mode plot is looking better now.
 

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Whats the best plan off attack for using this:
Do I use the REW EQ suggested filters when I carry out the 'match response target', is there a limit to how many I should use, I guess fewer is better, but what if I used all twelve.
As John discussed, make sure your Target Curve is properly set before you start equalizing. If you do that, it’ll only take a few filters. You’re mainly wanting to eliminate the worst peaks and depressions; there’s little (if any) audible benefit to smoothing out every little ripple in response.

I've also read in the threads about not adding to much Gain above 3dB.....?
Yeah, it’s one of those myths that just won’t go away, and it’s based on a number of flawed theories.

One misconception is based on the premise that boosted filters will require the input signal of the digital equalizer to be cut back, and that will decrease dynamic range and increase noise in the A/D converter. This is an old-school idea that dates back to the days of 16-bit digital converters.

The high-input-levels-required theory confuses the unit’s overall dynamic range with the dynamic range of the A/D converter itself. The two are not the same. A 24-bit A/D converter has a theoretical dynamic range of 147 dB. However, other circuitry in the component also contributes to the component’s ultimate S/N and dynamic range spec – the power supply, balancing ICs, etc. The reason why no 24-bit digital processor is able to spec an actual 147 dB dynamic range is because the analog circuitry in the device can’t deliver that kind of performance.

If a 24 bit A/D converter has a theoretical dynamic range of 147 dB, is reducing the input level 10 or 15 dB going to have an appreciable affect on its dynamic range? Of course not – not when the equalizer itself only has a dynamic range of 105 dB or less. Bottom line, modern 24-bit processors function just like analog processors, and input levels are a moot issue – hence the reason they don’t have the handy front-panel level controls they used to.

Another myth says that you shouldn’t boost dips or nulls in response because it doesn’t accomplish anything. We’re told that nulls are caused by phase cancellations, where the sound reaching the measurement point is a combination of the original direct soundwave and a reflected soundwave that is 180˚ out of phase (one half wavelength). When you add a gain filter at the depressed frequency, not only does the direct sound increase by the number of dB of the filter, but unfortunately the 180˚ out-of-phase signal also applies an equal and opposite signal to counteract. The result is that your dip is still there and you have wasted the gain you've thrown at it.

This theory is technically correct, but it’s based on the silly expectation that bass emanates from a subwoofer like a laser, with the primary “sound beam” reflecting off a perfectly-perpendicular wall and meeting the original “sound beam” half-way in between, 180˚ out of phase, causing cancellation.

Of course, the theory falls apart with the realization that bass frequencies are omnidirectional, not directional. They discharge from the subwoofer in all directions and in like manner bounce around the room in all directions. This quote from this article explains things nicely:

These nulls are related not only to the distance from the rear wall, but also from the other walls, the floor, and the ceiling. So in order to create a deep null a precise balance is needed, and that balance is easily disrupted by the many other reflections bouncing around the room.

In other words, since bass is omnidirectional, sound is reflecting at every angle from every boundary in the room. So while there may well be a point and/or frequency that exhibits wholesale cancellation due to a reflection being 180˚ out of phase with the original signal, there is no shortage of other, non-canceled reflections present for an equalizer to work with. Plus, the EQ introduces phase changes itself. This is why you’ll seldom see a case where a depression won’t respond to EQ.

The other problem with the “never boost nulls” theory is that it presumes all depressions, dips, troughs etc. in response are nulls. The explanation above should be sufficient to show that true nulls are pretty rare. They’re usually the result of a poor subwoofer location.

It’s pretty easy to identify nulls: Typically they are deep and narrow and won't respond to equalization, as we see with these graphs:


example of unequalizable null.jpg

Graph with nulls after ss.jpg


Here’s the second graph, but with EQ. As you can see, the nulls didn’t budge.


Graph with nulls after EQ.jpg


By contrast, here’s a graph of a sub with a rather severe depression that boosted just fine with equalization. Obviously it wasn’t a null.


M&K baseline.jpg

M&K equalized.jpg


Another “no boosting” myth says that boosting enacts a headroom penalty, because it places increased demands on the amp and driver. In reality, virtually any equalization always enacts a headroom penalty, even if cut-only filters are used.

The common situation is that there are one or two peaks in the room. Those hot frequencies are what’s determining the volume setting you’re using for the sub before equalization. Eliminate (cut) those peaks and you’ll now perceive that your sub is too quiet. So, you have to turn it up to compensate. Well guess what? Now you’re driving your sub harder than you were before.

This is exactly what happened to the little 8” M&K sub I used to have for my computer audio system. It required some substantial EQ cuts to smooth it out – no boosts, only cuts. Sounds much better, of course, but the 10-11 dB of SPL lost to the cuts required that I had to increase its gain. Well - now it doesn’t take much for it to bottom out.

Also, consider what happens when you make a series of cuts. It shouldn’t take much to figure out that you now have peaks in between them!

For instance, consider the graph below, which depicts the equalizer’s electronic response after equalizing. In other words, the equalization changes the electrical signal that was originally flat (i.e. the flat blue line, which also includes the slope of the crossover). So, did we make three cuts here...




...or did we make three boosts, one being a power-robbing shelving filter?




See what I mean?

Bottom line, there’s no free lunch. In most cases, equalization requires that you have ample headroom going it, because you’re going to end up with less than what you started with. If you don't have ample headroom, you shouldn’t equalize.

It also should be mentioned - for those with ported subs, it’s not good to boost below the tuning frequency of the port. It will run out of headroom much sooner than a sealed sub with low frequency boost will.

Regards,
Wayne
 

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Discussion Starter #33
Thanks for your help chaps :T

'I'm getting fed up of moving my sub around, it's got to come to an end soon!'
 

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Discussion Starter #34
Right, I've added my REL storm in to the mix to see if this would improve things. After trying all different settings etc I found that the REL out of phase by 180 degree's with the SVS gave best results.
However, when I added my main speakers to the mix I found I had to have them running at the same phase, also leaving my mains at 80hz cut off so far worked best.

My question is before I spend all day tomorrow messing around trying to get this set up should I leave the REL out, or is it a case if it works leave it in ?? :scratch:
 

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Discussion Starter #35
Don't think you need to answer. I've just listened to some music and it sounded awful, really boomey.
It did sound a lot better when I reversed the phase on the REL......

But think I will try and set up my system without it, it will make things alot easier.
 

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Discussion Starter #36
Sorry, was meant to say:

First pic:
SVS-phase 0, REL-phase 180, Main spk's set to 80hz

Second pic:
I ran and EQ on it to see what it predicted.

Third pic:
I ran EQ on SVS-phase 0, REL-phase 180 only.

Could a BFD be used for both sub's if used before the Anti-mode?
 

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When the mains are active you need to change the target curve to "full range", the subwoofer target is for use when only the sub was active during the measurement. Also click on "Set Target Level" to get REW to adjust the level to better track the response, the target level is too low in the plots you posted.
 

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Discussion Starter #38
Well I've been testing all day and I am not moving that sub no more. I found the sub sitting in line with my main speakrs gave the best result, the Misses will just have to put up with looking at it!

The results are 'hope I've done them correctly this time'

1 sub-anti-mode and BFD used
2 sub EQ ' 10 filters used'
3 Main spk's 120hz, sub 80hz Anti BFD used
4 Main spk's 80hz, sub 80hz Anti BFD used.

So far what I have listened to sound really good. The BFD has tightened the Bass up nicely.
 

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